The invention relates, inter alia, to a method for forwarding signaling data in an interworking unit, with the operation:                Receiving data in an interworking unit or a gateway from a first data transmission network (CS) over a data transmission connection or over a bearer connection, in which, for a data transmission service between two mobile stations (MS1, MS2) or a group of stations, signaling data and payload data are transmitted, with the interworking unit transmitting payload data between the first data transmission network (CS) in which signaling is in accordance with a first signaling method and a second data transmission network (IMS) in which signaling is in accordance with a second signaling method, with the first signaling method differing from the second signaling method.        
In addition to the so-called “Circuit Switched (CS) domain” of a mobile radio network based on the 3rd Generation Partnership Project (3GPP), the so-called “IP Multimedia Subsystem” (IMS) is used for voice and video telephony and a so-called “interworking” of the relevant services, i.e. a connection of the services by a suitable conversion of the signaling used and of the bearer format of the data used is necessary between IMS and CS domain. As well as being used for the 3GPP “Global System for Mobile Communications” (GSM) and “Universal Mobile Telecommunications System” (UMTS) access networks, the IMS is also used for other access networks, for example “Wireless Local Area Network” (WLAN) and “Digital Subscriber Line” (DSL). It is precisely in these scenarios that it is initially to be expected that voice and video telephony will be undertaken via the IMS. Video telephony can also be used in a public telephone network, i.e. a Public Switched Telephone Network (PSTN), with the same in-band video-telephony-specific protocols being used as a rule for transport and signaling as in the 3GPP CS domain. Interworking from the PSTN to the IMS is also necessary.
Previously the standard has merely described interworking between IMS and CS domain or PSTN for voice telephony only. The present invention relates to the appropriate interworking for other services, especially for multimedia services, for example for video telephony. A demand for this is to be foreseen, since video telephony is increasing in significance both in the 3GPP CS domain and also in IMS, here in particular for access networks such as WLAN or DSL, or newly-arising network access options (e.g. Worldwide Interoperability for Microwave Access (WiMAX).
The interworking between IMS and a CS network, i.e. a PSTN or a 3GPP CS domain, is specified in 3GPP TS 29.163 from 3GPP Release 6 onwards only for pure voice telephony. In accordance with TS 29.163, the interworking of what is known as the call-control signaling takes place in the Media Gateway Control Function (MGCF). The interworking of the payload connection, i.e. the onward transfer and repackaging as well as if necessary the transcoding of the payload data, is undertaken in the so-called Internet Multimedia-Media Gateway (IM-MGW). The MGCF controls the IM-MGW by the H.248 protocol standardized by the ITU-T via the Mn interface, as further described in 3GPP TS 29.332.
In the CS network Bearer Independent Call Control (BICC), see ITU-T (International Telecommunication Union-Telecommunication Standardization Sector) Q.1902.x, or ISDN User Part (ISUP), see ITU-T Q.761 ff, is used for out-of-band call control signaling. In the case in which the call control signaling is routed separately from the bearer connections, this method is also referred as out-of-band signaling. Subsequently there also the option within the bearer connection of exchanging signaling messages, which is referred to as in-band signaling. In the case of ISUP, Time Division Multiplex (TDM) is used as bearer in the CS network, and in the case of BICC packet transport by Internet Protocol (IP) or Asynchronous Transfer mode (ATM). The negotiation about whether pure voice telephony or video telephony are used can be undertaken for ISUP during the call control signaling for setting up the call by the so-called ISUP UDI Fallback procedure. For BICC this negotiation can occur by means the Service Change and UDI Fallback (SCUDIF) standardized in 3GPP TS 23.172, which also allows a change between voice telephony and video telephony during a call. Both UDI Fallback and SCUDIF use out-of-band signaling. In addition it is possible both for ISUP and BICC to not use the procedure and only attempt a call setup for video telephony, and, in the event of video telephony not being supported, abort the call setup. By contrast with optional negotiation between voice and video the negotiation of the voice and video codecs used for video telephony is undertaken “in-band”, after video telephony has already been selected beforehand and a corresponding bearer connection has been established. A so-called BS30 data connection with a bandwidth of 64 kbyte/s is used for video telephony in the network. Within this data connection the H.324 protocol suite standardized by the ITU-T is used, with the variant H.324M adapted for mobile telephony being selected in the 3GPP CS domain. After the data connection is set up in this case the configuration of the multimedia connection is negotiated in-band via the ITU-T standardized H.245 protocol, in particular the video codec and speech codec used and the details of the respective codec configuration Voice and video as well as the signaling data are multiplexed by the H.223 protocol in the same bearer connection. For the 3GPP CS domain TS 26.110 further describes the use of the H.324 protocol suite or protocol series, with especially the so-called H.324M configuration being selected.
The most important execution sequences in setting up a 3G-324M session are as follows:
1. After the start of the ISUP or BICC call setup signaling, necessary resources are reserved that are needed for the desired “bearer” and subsequently the bearer is set up.
2. Start of the “in-band” negotiation. Initially negotiation of the H.223 multiplexer level which is to be used for this bearer.
3. Recognition of the master station which is opening the multistream connection by H.245 negotiation if necessary. This function is only needed if a conflict arises within the context of opening a bidirectional logical channel. This function is referred to as Master or Slave Determination (MSD).
4. The capabilities of the station sending the message are transmitted by so-called “Terminal Capability Set” H.245 messages. Such messages are sent independently of the two stations. These described capabilities contain the following information: Audio and video codec and their specific characteristics or their variants. Functional scope of the multiplexer, in detail which adaptation layer is supported (e.g. simple or nested multiplexing) and its mobile-specific extensions.
5. Setting up of “logical” channels for each media stream by H.245 signaling. From this point in time onwards, either with MSD or without, the station or the IM-MGW are ready to open logical channels to allow the exchange of voice, and/or video payload data. In the creation of a bidirectional logical channel, the channel number and the final media capabilities to be used are defined.
6. Definition of the multiplex characteristics by H.245.
7. Start of the transmission of video, audio/voice or data
Negotiation for video telephony is undertaken “out-of-band” in the IMS with the aid of the Session Description Protocol” (SDP), IETF (Internet Engineering Task Force) RFC (Request for Comment) 2327, which is transported by the Session Initiation Protocol (SIP), IETF RFC 3261. In this case the negotiation as to whether voice telephony or video telephony is used in linked to the negotiation of the codec used and is undertaken before or during of the setting up of the bearer. The SDP offer-answer mechanism in accordance with RFC 3264 is used. In this case the offering party sends a list of supported codecs in the SDP Offer message. After receiving this message the answering party sends an SDP Answer message containing the codec from the list that it also supports and wishes to use. The answering party may not specify any codecs that were not contained in the list of the SDP offer. By contrast with the CS domain, two separate bearers are used for voice and video, which each use the Real Time Transport Protocol (RTP), IETF RFC 3550. For the 3GPP IMS over the General Packet Radio Service (GPRS) access network 3GPP TS 26.235 describes the codecs to be used for video telephony.
Summarized below once again are the protocols and codecs used on the CS domain side and on the IMS side for video telephony.
CS network (especially 3GPP CS domain):
Call Control: BICC or ISUP.
Negotiation between pure voice telephony networks and video telephony can be undertaken for ISUP by UDI Fallback and for BICC by SCUDIF.
Multimedia Protocol suite: ITU-T H.324M (ITU-T H.324 Annex C)
Codec negotiation: ITU-T H.245 in-band negotiation about the CS bearer set up with 64 kbit/s (kilobits per second)
Video codec: Support of H.263 prescribed
ITU-T H.261 optional
MP4V-ES (simple video profile level 0) optional
Speech codec: Support of NB-AMR (Narrow Band Adaptive MultiRate) prescribed
WB-AMR (Wide Band AMR) optional
ITU-T G.723.1 recommended
Transport Multiplexing of voice and video in a bearer in accordance with ITU-T H.223 Annex A+B
IMS (codecs for GPRS (General Packet Radio Service) access network)
Call Control: SIP
Includes both negotiation between pure voice telephony networks and video telephony, and also codec negotiation.
Codec negotiation: Before setup of the bearer, out-of-band by SDP, which is transported in SIP.
Video codec: Support of H.263 prescribed
ITU-T H.264 optional,
MP4V-ES (simple video profile level 0) optional
Speech codec: Support of NB-AMR and WB-AMR prescribed.
Transport Two separate RTP bearers for voice and video using different so-called RTP Payload” formats:
Voice NB-AMR+WB-AMR: IETF RFC 3267
Video: H.263: IETF RFC 2429
H.264 (AVC): IETF RFC 3984
MPEG-4: IETF RFC 3016 parallel RTP media streams are synchronized by RTP timestamps which are negotiated by the Real Time Control Protocol (RTCP, see IETF RFC 3550).
As well as or in place of the codec specified here, other codecs can also be supported by the stations, especially if the CS stations are located in the PSTN or the IMS stations do not use GPRS as the access network.
With interworking for exclusively voice telephony out-of-band signaling is used in both networks. A conversion of the signaling protocol can thus be comparatively easily performed at the borders of the two data transmission networks because all signaling messages can be merged in a simple manner at one unit. On the other hand in-band signaling is used in the CS network with video telephony for example which is received by the IM-MWG, whereas in the IMS out-of-band signaling is used which is received by the MGCF.
However the problem of merging the signaling between the two data transmission networks also occurs with other multimedia services or with other services.